Adaptive and progressive scrambling of audio streams

ABSTRACT

A process for distributing digital audio sequences according to a nominal stream format that include a succession of frames, each of which includes at least one digital audio block grouping a plurality of coefficients corresponding to digitally coded audio elements, including modifying at least one block of an original stream of sequences, in an adaptive manner on the original stream as a function of at least a part of characteristics representative of the structure, content and parameters of the original stream, a target profile and external events.

RELATED APPLICATION

This is a continuation of International Application No.PCT/FR2003/050099, with an international filing date of Oct. 21, 2003(WO 2004/039053, published May 6, 2004), which is based on French PatentApplication No. 02/13091, filed Oct. 21, 2002.

FIELD OF THE INVENTION

This invention relates to processing digital audio streams. Moreparticularly, the invention relates to a system permitting the auditoryscrambling and recomposing of digital audio content.

BACKGROUND

WO 00/58963 (Liquid Audio) discloses that data such as a musical trackis saved as a secure portable track (SPT) that can be linked to one orseveral players and can be linked to a particular saving means, thusrestricting reading the SPT to specific players and ensuring that thereading is carried out only from the original saving means. The SPT islinked to a player by the encryption of data of the SPT using a save keythat is unique to the player, difficult to change and is guarded by theplayer under strict security conditions. The SPT is linked to aparticular means of saving including data uniquely identifying the savemeans in a form resistant to falsification, that is, signed in anencrypted manner.

A system for scrambling audio signals is also known from U.S. Pat. No.4,600,941 (Sony) in which an audio signal is divided into blocks, eachof which is formed by a plurality of frames, which plurality of framesis rearranged on a time base in an order predetermined for each block insuch a manner as to be encoded. The encoded signal is rearranged on atime base in an original order in such a manner as to be decoded. Thissystem comprises a first circuit for processing the signal to insert aredundant portion into a portion between contiguous frames and tocompress the frames in base time in response to the redundant portionsduring the encoding, a circuit generating a signal for inserting acontrol signal other than audio information in the redundant portions, acircuit for detecting the control signal for detecting the controlsignal during decoding and a second circuit for processing the signalfor removing the redundant portions in synchronism with the detectedcontrol signal and decompressing the frames in base time in response tothe redundant portions.

A method and a system for scrambling and descrambling audio informationsignals is also known from U.S. Pat. No. 5,058,159 (Macrovisioncorporation). The audio signals are scrambled by inverting the originalfrequency spectrum in such a manner that the frequency portions that areoriginally at the bottom in the audio frequency band are shifted to thetop, whereas the portions originally at the top of the band are shiftedto the bottom. A pilot sound of a known frequency is recorded with theaudio signals of the shifted frequencies. During reproduction, eachvariation in phase and in frequency is searched by its pilot that usedto generate the modulation signal for reconstituting the originalcontent in audio signal frequencies.

WO 99/55089 “Multimedia Adaptive Scrambling System” discloses a systemfor scrambling digital samples representing multimedia data (audio andvideo) in such a manner that the content of the samples is degraded, butrecognizable or otherwise supplied with the required quality. The levelof quality is linked to an associated signal/noise ratio and determinedwith the aid of objective and subjective tests. A given number of LSB's(least significant bits) is scrambled frame by frame in an adaptivemanner as a function of the dynamics of the possible values. All theencryption keys are included in the audio/video stream and used by thedecoder for descrambling and restoring the stream. After thedescrambling the encryption key cannot be recovered because it isscrambled itself by the decoder.

The art provides evidence of a number of systems for protecting audiostreams based substantially on the encryption of data, adding encryptionkeys independent of the content of the audio stream and which thereforemodify the format of the structured stream. One particular and differentrealization is that of the Coding Technologies company, that consistsprotecting by scrambling a selected part of the bitstream (“bitstream”refers to the binary stream at the output of the audio encoder) and notthe entire bitstream. The protected parts represent the spectral valuesof the audio signal with the result that during the decoding withoutdecryption the audio stream is distorted and disagreeable to the ear.

SUMMARY OF THE INVENTION

This invention relates to a process for distributing digital audiosequences according to a nominal stream format that includes asuccession of frames, each of which includes at least one digital audioblock grouping a plurality of coefficients corresponding to digitallycoded audio elements, including modifying at least one block of anoriginal stream of sequences in an adaptive manner on the originalstream as a function of at least a part of characteristicsrepresentative of the structure, content and parameters of the originalstream, a target profile and external events.

This invention also relates to a system for distributing digital audiosequences including an audio server including means for broadcasting astream modified in conformity with the process for distributing digitalaudio sequences and a plurality of pieces of equipment provided with ascrambling circuit, wherein the server includes means for recording thedigital profile of each target and means for control of the modificationmeans as a function of input variables corresponding to at least a partof the characteristics representative of the structure, the content andthe parameters of the original audio stream, the target profile and theexternal events.

BRIEF DESCRIPTION OF THE DRAWING

The Drawing schematically shows a client system in accordance withaspects of the invention.

DETAILED DESCRIPTION

The invention addresses the disadvantages of the prior art by providingan adaptive and progressive scrambling as a function of the structure ofthe audio stream, of the client profile and of external events.

The invention thus relates to a device capable of transmitting in asecure manner a set of audio streams with a high auditory quality to amusical or speech player to be recorded in the memory or on the harddisk of a set-top decoder box connecting the transmission network to theaudio player while preserving the auditory quality, but avoidingfraudulent use such as the possibility of making pirated copies of audioprograms recorded in the memory or on the hard disk of the set-topdecoder box.

The invention also relates a process for distributing digital audiosequences according to a nominal stream format constituted of asuccession of frames, each comprising at least one digital audio blockgrouping a certain number of coefficients corresponding to simple audioelements coded digitally according to a manner specified in the streamconcerned and used by all audio decoders capable of playing it to beable to correctly decode it. This process comprises:

-   -   a preparatory stage comprising modifying at least one of the        coefficients,    -   a transmission stage    -   of a main stream in conformity with the nominal format        constituted of frames containing the blocks modified in the        course of the preparatory stage and    -   by a path, separate from this main stream, of complementary        digital information allowing the reconstitution of the original        stream from the computation on the target equipment as a        function of the main stream and of the complementary        information. The complementary information is defined as a set        constituted of data (e.g., coefficients describing the original        data stream or extracts of the original stream) and of functions        (e.g., the substitution or interchanging function). A function        is defined as containing at least one instruction putting data        and operators in a relationship. This complementary digital        information describes the operations to be carried out for        recovering the digital stream from the modified stream.

Reconstitution of the original stream is carried out on the targetequipment from the modified main stream already present or sent in realtime on the target equipment and from the complementary information sentin real time at the moment of listening and comprising data andfunctions executed with the aid of digital routines (set ofinstructions).

The term “scrambling” denotes modification of a digital audio stream byappropriate methods in such a manner that that the stream remains inconformity with the norm or standard with which it was digitally encodedwhile rendering it audible by an audio reader (or player), but alteredas concerns human auditory perception.

The term “descrambling” denotes the process of restoration byappropriate methods of the initial stream and the restored audio streamis identical after descrambling to the original initial audio stream.Reconstitution of the original stream is carried out on the targetequipment from the modified main stream already present or sent in realtime on the target equipment and from the complementary information sentin real time at the moment of listening and comprising data andfunctions executed with the aid of digital routines (set ofinstructions). The entirety or a subpart of the complementaryinformation is sent as a function of the profile and of the rights ofthe client. The quantity of information contained in this subpart of thecomplementary information is defined as the number of data and/orfunctions belonging to the complementary information sent to the targetduring the connection.

The type of information contained in this subpart corresponds to a levelof scalability determined as a function of the profile of the target.The nature of the data and/or functions belonging to the complementaryinformation sent to the target during the connection is defined as thetype. For example, the type of data is relative to the habits of thetarget (connection time, duration of the connection, regularity of theconnection and of payments), to the environment (lives in a big city,the time at the present moment) and to personal characteristics (age,sex, religion, community).

Complementary information is composed at least of functions that arepersonalized for each target relative to the connection session. Asession is defined starting from the connection time, the duration, thetype of the modified stream listened to and the connected elements(targets, servers).

The complementary information is subdivided into at least two subparts,each of which can be distributed by different media or by the samemedium. For example, in the case of distribution of the complementaryinformation by several media a more complex management of the rights ofthe targets can be ensured.

The term “profile” of the user denotes a data file comprisingdescriptors and information specific to the user, e.g. culturalpreferences and social and cultural characteristics, habits of use suchas the frequency of using audio means, the average listening time of ascrambled audio sequence, the frequency of listening to a scrambledsequence, the price the user is ready to pay or any other behavioralcharacteristic regarding the use of audio sequences. This profile isformalized by a data file or a data table that can be used by computermeans.

Many scrambling systems have an immediate effect in that the initialstream is totally scrambled or the initial stream is not scrambled atall. Also, generally different audio sequences can be scrambled with thesame algorithm and the same regulating parameters. Numerous protectionsused do not change the scrambling of an audio stream as a function ofits contents.

An adaptive and progressive scrambling is supplied as a function of thestructure of the audio stream (bitstream) and/or of its contents whilechanging the algorithms and the parameters of the scrambling as afunction of the characteristics of the audio stream and of the userapplication to realize reliable protection regarding deterioration ofthe original stream and resistance to pirating at a minimum cost andassuring the quality of service required by the target or the client.Various adaptations of scrambling are applied, e.g., like those citedbelow.

The invention concerns in its most general meaning a process fordistributing digital audio sequences according to a nominal streamformat that are constituted a succession of frames, each of whichcomprises at least one digital audio block grouping a plurality ofcoefficients corresponding to simple, digitally coded audio elements,which process comprises a stage for modification of at least one blockof the original stream, characterized in that this modification stageacts in an adaptive manner on the original stream as a function of atleast a part of the characteristics representative of the structure, thecontent and parameters of the original audio stream, the target profileand external events.

The modification stage preferably comprises replacing a part of thecoefficients to produce on the one hand a main audio stream in nominalformat and on the other hand complementary modification information thatallows the reconstruction of the original stream by a decoder of thetarget equipment, the scope of which modifications is variable anddetermined by the representative characteristics.

The modified main stream may be recorded on the target equipment priorto the transmission of the complementary information on the targetequipment. The modified main stream may also be recorded on a physicalsupport to be transmitted to the target equipment prior to transmissionof the complementary information on the target equipment. The modifiedmain stream and the complementary information may be transmittedtogether in real time.

This complementary modification information advantageously comprises atleast one digital routine suitable for executing a function. Thecomplementary modification information may be subdivided into at leasttwo subparts. The subparts of the complementary modification informationmay also be distributed by different media. The subparts of thecomplementary modification information may be distributed by the samemedia. The complementary information may further be transmitted on aphysical vector. Finally, the complementary information may betransmitted online.

The digital audio sequences are advantageously modified in adifferentiated manner as a function of their audio content. The digitalaudio sequences are advantageously modified in a differentiated manneras a function of the layer of modified scalability. The digital audiosequences are also advantageously modified in a differentiated manner asa function of the rate in kilobits per second (kbits/s) of the originalstream.

The digital audio sequences may be modified in a differentiated manneras a function of the profile and the digital level defined by the normor the standard with which they were encoded. The digital audiosequences may also be modified in a differentiated manner as a functionof the number of audio channels present in the stream.

The digital audio sequences are advantageously modified in adifferentiated manner as a function of coupling and multiplexing betweendifferent audio channels present in the stream. The digital audiosequences may be modified in a differentiated manner as a function ofthe sampling frequency with which the audio stream was encoded. Thedigital audio sequences may be modified in a differentiated manner as afunction of the psychoacoustic model used. The digital audio sequencesmay further be modified in a differentiated manner as a function oftheir granular scalability.

The digital audio sequences are advantageously modified in a progressivemanner increasing the degradation effect up to the complete scramblingof the audio stream. The digital audio sequences are preferably modifiedwith a random generation of the scrambling parameters andconfigurations.

The process preferably comprises a prior analog/digital conversion stagewith a structured format, which process is applied to an analog audiosignal.

The invention also relates to a system for distributing digital audiosequences comprising an audio server comprising means for broadcasting astream modified in conformity with any one of the preceding processesand a plurality of pieces of equipment provided with a scramblingcircuit, characterized in that the server also comprises means forrecording the digital profile of each target and means for controllingthe modification means as a function of input variables corresponding toat least a part of the characteristics representative of the structure,the content and the parameters of the original audio stream, of thetarget profile and of external events.

A digital audio stream is generally composed by sequences comprisingframes or blocks organized according to a digital format specific foreach audio coder, including the headers of the frames with the variousparameters of encoding and coefficients relative to a specificrepresentation of digital audio samples. Given knowledge of the mannerin which the modeling, compression and encoding of the audio signal forthe audio coder and/or the given standard or the norm are carried out,it is possible to extract the main parameters from the bitstream thatdescribe it and that are sent to the decoder.

Once these parameters are identified, they are modified in such a mannerthat that the audio stream generated by the given coder and/or standardis in conformity with the coder and/or standard. Moreover, themodification ensures stability of the sound signal, but renders itunusable by the user, because it is scrambled. Nevertheless, it can beunderstood and interpreted in the decoder corresponding to its encodingand played by a player without the latter being disturbed.

Modification of one or several of the components of the audio signal(spectral envelope, fundamental or harmonics, psychoacoustic model, timedivision development, signal/noise ratio, composition, compression,quantification, transformation) cause its degradation from an auditorystandpoint and transform it into a signal that is completelyincomprehensible as concerns the subjective auditory perception. Thepart of the audio signal or the component describing it that will bemodified depends on its encoding for each given coder-decoder regardlessof whether for speech, music, sound or special effects, synthetic soundsor any audio signal of the same type. Depending on the manner in whichencoding and transformation of the resulting parameters are realized, itis possible to have direct or indirect information about the maincharacteristics of the audio signal and thus modify them. This principleis applicable to all types of digital coders as well as to all theirbase and enhancement layers or the combination of both.

An adaptation of the scrambling parameters is applied as a function ofthe content of the audio stream: Natural or synthetic speech, music,noise, natural or synthetic or compound sounds, special effects. Forexample, the HVXC (harmonic vector eXcitation coding) encoder for speechand the HILN (harmonic and individual lines plus noise) for music,defined by the MPEG-4 norm, are parametric coders that code the audiosignal separately or conjointly as a function of its content.

For example, in the case in which speech is predominant the bitstreamcoming from the HVXC contains the values of the LSP (line spectralpairs) reflecting the LPC (linear predictive coding) parameters. Thevalues of the LSP of the current frame are quantified vectorially in twostages, stabilized in one value to ensure the stability of the LPCsynthesis filter and then arranged in a bitstream in ascending orderwith a minimum distance between adjacent coefficients.

The subscripts of the vectorially quantified LSP pairs are transmittedto the decoder that restores the values of the LSP and therefore of theLPC from standard tables. By replacing the original subscripts withother values taken from predefined tables in the norm the bitstream willremain in conformity, but the decoded LSP values will not correspond tothe original LPC parameters. As a consequence, the spectral envelopewill be modified and the speech deteriorated.

Many audio coders are characterized by scalability. The notion of“scalabilite” is defined from the English word “scalability”, whichcharacterizes an encoder capable of encoding or a decoder capable ofdecoding an ordered set of binary streams in such a manner as to produceor reconstitute a multilayer sequence. A scrambling that is adaptiverelative to the base layer or the enhancement layers is applied as afunction of the configuration of the audio encoder. For example, theHVXC and HILN encoders each possess a base layer and an enhancementlayer, which allows several possible configurations. The parameters forthe base layer, the enhancement layer or for the two layers are modifiedas a function of the degree of scrambling desired.

An adaptation is also applied as a function of the rate in number ofkilobits per second (kbits/s) of the audio stream whether it is constantor variable. For certain more complex audio streams (like those of theMPEG-4 type, that have a variable rate in very large proportions (from 2kbits/s to 64 kbits/s), the scrambling parameters are selected as afunction of the rate, given that the scrambling for a low rate on theorder of 2 kbits/s turns out to be less effective for higher rates wherethe encoding precision is much greater.

An adaptation of the scrambling parameters is also applied as a functionof the fine granular scalability, stemming from the English term “finegranular scalability” characterizing certain audio streams. The notionof “scalabilite granulaire” is defined from the expression in English“granular scalability” used in the MPEG-4 norm that characterizes anencoder capable of encoding or a decoder capable of decoding an orderedset of binary streams in such a manner as to produce or reconstitute amulti-layer sequence. Granularity is defined as the quantity ofinformation that can be transmitted per layer of a system characterizedby any scalability, which system is then also granular. For example, theAAC encoding scheme (advanced audio coding) with BSAC (bit slicedarithmetic coding) creates the possibility of an encoding with reductionof the noise of an AAC bitstream in a bitstream with a fine granularscalability between 16 kbits/s and 64 kbits/s per channel, of which thebinary rate can be modulated with a step of 1 kbits/s.

For certain more complex audio streams (like those defined by the MPEG-4norm) an adaptive scrambling is applied as a function of the types ofobjects contained in the stream, of the profile, level designating thecomplexity and the options used during construction of the audio stream.In fact, there are a multitude of objects and of audio profiles in theMPEG-4 audio framework. For example, for the natural audio objects, oneof the profiles is the simple scalable one that contains the CELP (codeexcited linear prediction) tools and AAC (advanced audio coding).Scrambling is carried out as a function of the parameters of these twocoders. Adaptive modification of the elements of the audio stream iscarried out as a function of the types of audio objects that eachprofile and level contain. An adaptation of the scrambling parametersmay also be applied as a function of the number of audio channelspresent in the stream. An adaptation of the scrambling parameters may beapplied as a function of the coupling and of the multiplexing betweenthe various audio channels present in the stream. An adaptation of thescrambling parameters may further be applied as a function of thesampling frequency with which the audio stream was encoded. Anadaptation of the scrambling parameters may be applied as a function ofpsychoacoustic model used characterizing certain audio encoders.

For example, in the AAC MPEG-4 norm, the psychoacoustic model estimatesthe thresholds determining the maximum quantification error that can beadmitted during compression while preserving the audio quality. Thespectral data is quantified and coded as a function of these estimatedthresholds. Quantification is selected as a function of the estimatedthresholds, e.g., quantification can be uniform or non-uniform and it iscarried out with the aid of scale factors. By modifying the values ofthese scale factors coded in differential in the binary stream, aquantification error is introduced because the scale factors no longercorrespond to those defined by the estimations of the psychoacousticmodel. Scrambling is adapted as a function of the desired auditorydegradation. In a case in which a slight scrambling would be desired thelast scale factors are modified. It is advantageous if a strong auditorydegradation is desired that the first scale factor is modified. Giventhat all the scale factors are coded in differential relative to thefirst scale factor all the values that follow are erroneous and theaudio signal is strongly disturbed.

A progressive scrambling is also applied in such a manner that the userbegins to hear the non-scrambled audio stream. Then, a slight scramblingis begun that is reinforced more and more until the audio stream becomesentirely scrambled. The goal striven for is to awaken the interest ofthe user for the audio stream, but to remove the rights to hear it ifthe user did not purchase them. A realization of this application is toscramble the audio stream with one or several of the given algorithmswhile progressively modifying the scrambling parameters during a timedetermined in such a manner as to increase the unpleasantness untilarriving at a completely scrambled and inaudible stream.

An adaptive scrambling is generally realized as a function of thecontent, characteristics, structure and composition of the digitalstream defined by a norm or a given standard. Scrambling is alsorealized with a random generation of parametric combinations to beapplied to scramble the audio stream. A protection that is robust anddifficult to attack or that can not be pirated by an ill-disposed personis ensured in this manner.

An adaptation of the scrambling parameters and algorithms is alsoapplied as a function of the target profile, as a function of the targetbehavior during the connection to the server (e.g., the regularity andsubmission of payments), as a function of the price that be paid, as afunction of habits (e.g., time, time of connection), as a function ofcharacteristics (e.g., age, sex, religion, community), or as a functionof data communicated by a third party (belonging to associations orpresent in consumer databases).

An adaptation of the scrambling parameters and algorithms is alsoapplied as a function of external events as, e.g., the broadcastingtime, audience rate, sociopolitical events or disturbances during thebroadcasting.

The invention will be better understood with the aid of the followingdescription made purely by way of explanation of a selected aspect ofthe invention with reference made to the drawing which shows aparticular agent of the client-server system in accordance with theinvention. The audio stream of the MPEG-AAC type that is to be secured 1is sent to an analyzing 121 and scrambling 122 system that generates amodified main stream and complementary information at the output.

The original stream 1 can be directly in digital form 10 or in analogform 11. In the latter case, analog stream 11 is converted by a coder(not shown) in digital format 10. In the remainder of the text we willtake note 1 of the input digital audio stream.

A first stream 124 in the MPEG-AAC format with a format identical to theinput digital stream 1 except for the fact that some of its coefficientsand/or values have been modified, is placed in an output buffer memory125.

The complementary information 123 in any format contains the referencesto the parts of the audio samples that are modified and is placed inbuffer 126. The analysis 121 and scrambling 122 system decides as afunction of the characteristics of input stream 1 which adaptivescrambling to apply and which parameters of the stream to modify andalso, as a function of the rights of the client, in which manner toapply the modifications, e.g., progressively or not.

The MPEG-AAC stream 125 is then transmitted either in physical form on aCD-ROM, non-volatile memory, DVD, or the like or via a network 4 of thetelephone network type, DSL (digital subscriber line), BLR (local radioloop), DAB (digital audio broadcasting), RTC (commutated telephonenetwork), digital mobiles (GSM, GPRS, UMTS), microwave, cable,satellite, e.g., to the client 8 and more precisely into memory 81 ofthe RAM, ROM, hard disk type. When target 8 requests to hear an audiosequence present in memory 81, there are two possibilities:

-   -   Either the target 8 does not have the rights necessary to play        the audio sequence. In this case, stream 125 generated by the        scrambling system 122 present in memory 81 is passed to        synthesis system 82 that does not modify it and transmits it        identically to a classic audio player 83 and its contents,        heavily degraded auditorily, is played by player 83 on a headset        or on loudspeakers 9, or    -   Target 8 has the rights to hear the audio sequence. Server 12        transmits appropriate complementary information 126 as a        function of the rights of the target by connection 6        corresponding to the type of scrambling carried out. In this        case, the synthesis system makes a hearing request to server 12        containing the information 126 necessary to recover original        audio sequence 1. Server 12 then sends complementary information        126 by connection 6 via transmission networks of the following        types: analog or digital telephone line, DSL (digital subscriber        line), BLR (local radio loop), DAB (digital audio broadcasting),        RTC (commutated telephone network), digital mobile networks        (GSM, GPRS, UMTS), microwave, cable or satellite which        information permits the reconstitution of the audio sequence in        such a manner that the target 8 can hear and/or store the audio        sequence. Synthesis system 82 then proceeds to descramble the        audio sequence by reconstructing the original stream by        combining modified main stream 125 and complementary information        126. The audio stream obtained in this manner at the output of        synthesis system 82 is then transmitted to classic audio player        83 and the original audio sequences played on a headset or        loudspeakers 9.

The invention will now be described with the aid of a second exemplaryaspect showing modifications differentiated as a function of the rate,structure, composition of the audio frame and also as a function of theeffect of the auditory degradation to be obtained.

More and more coders have the option of functioning with variable ratesto satisfy specific applications as, e.g., to respond to the constraintsof limited bandwidth. An example of a coder designed to ensure anacceptable quality of speech while respecting a bandwidth with a lowrate is the AMR (“adaptive multi-rate” in English) coder, designed forcellular telephony that can function in eight different modes and whoserate varies between 4.75 kbits/s and 12.2 kbits/s. The invention carriesout modifications differentiated as a function of the mode with whichthe audio stream was encoded, that is, as a function of the rate, of thelength of the prospective components of the frame as well as a functionof the desired degree of auditory degradation.

For example, in the 12.2 kbits/s mode the structure of the AMR frame isthe following:

-   -   The subscripts corresponding to the spectral frequency pairs,        called LSF's (“line spectral frequencies”), relative to the        LSP's (“line spectral pairs”) parameters, therefore also to the        LPC (“linear predictive coding”), that is, to the form of the        filter of the formants, which subscripts are common to the        entire frame;    -   Four groups of parameters relative to four subframes contained        in the complete frame and representing one hundred and sixty        audio samples.

Each group of parameters per subframe is constituted in the followingmanner:

-   -   Delay of the fundamental (“pitch delay”),    -   Amplitude of the fundamental (“pitch gain”),    -   Data concerning the sign and the frequency position of the        excitation impulses,    -   Subscript relative to the gain of the table of values        (“codebook”).

These parameters are modified in a differentiated manner as a functionof the desired auditory degradation. For example, modifying the value ofthe delay of the fundamental by substitution with a different valuecauses a frequency offset: A lower value causes a deformation of thevoice and the effect obtained is a muffled sound with cracklings similarto an “extinction of the voice”. Modifying the amplitude of thefundamental by substituting it with a larger value causes a jerkydeformation, some parts are amplified and others “smothered”.

Several modifications also carried out on the values of the LSF's:

-   -   Substituting the values of the LSF's by fixed values produces a        known sound effect similar to a jammed radio channel;    -   Substituting the values of the LSF's by randomly changing the        subscripts entirely breaks the sound because this adds        cracklings of different frequencies and amplitudes producing a        very unpleasant sound and the speech becomes unintelligible;    -   By modifying one LSF the audible degradation is similar to a        noise of a “whistling” type, but a part of the sound remains        perceptible. In this case, modifications are adapted, e.g., for        pre-hearing applications (“teasing”) when it is desired that the        user can perceive the sound and choose to request the rights for        it or not. For example, an LSF is modified and modifications are        progressively added on the second LSF, the third, the fourth and        the fifth until the values of all the LSF's have been modified        by substituting the value of the subscripts with one and the        same value, for example. The result obtained in this case is the        concentration of the spectrum around a frequency, e.g., if the        subscripts are placed at one, an unintelligible, low-frequency        sound is obtained.

The differentiated modifications of the LSF's yield low-volumecomplementary information for a significant auditory degradation. Theyare preferably combined with other modifications.

The signs of the pulsations relative to the construction of theexcitation are advantageously modified. Furthermore, by substituting theposition of the pulsations with “false” positions, the excitation isalso modified and the sound is totally deformed.

For a 7.95 kbits/s mode the structure of the frame is similar exceptthat it contains a single set of three LSF's. Differentiatedmodifications are then applied taking this particularity into accountand the frame length corresponding to this mode.

For the other modes of the AMR coder the frame structure is slightlydifferent. It does not contain the amplitude of the fundamental nor thegain of the fixed value tables, but rather a set of gains relative tothe fixed and adaptive value tables used for scaling the excitationconstructed from the addition of the adaptive code-vectors and frominnovation. The modification supplied take account of thesespecificities. Modifying the LSF's produces a significant degradation;however, given that the audio rates are not very elevated, smallmodifications are sufficient for obtaining a strong auditorydegradation.

The differentiated modifications are preferably carried out takingaccount of the rate desired for the complementary information.

The invention is not limited to the selected aspects cited as exemplaryembodiments, which modifications guarantee that the authorized amplitudevalues of the sound are not exceeded and guarantee the conformity of themodified main stream with the original audio stream.

It is advantageous if, after reconstitution on the equipment of the userfrom the modified main stream and from the complementary information,the reconstituted stream is auditorily identical to the original, butdifferent from a binary standpoint from the original stream to reinforcethe security.

It is advantageous if, after reconstitution on the equipment of the userfrom the modified main stream and from the complementary information,the reconstituted stream is strictly identical to the original and theprocess is without loss.

1. A process for distributing digital audio sequences according to anominal stream format that are comprised of a succession of frames, eachof which comprises at least one digital audio block grouping a pluralityof coefficients corresponding to digitally coded audio elements,comprising: modifying at least one block of an original stream ofsequences, in an adaptive manner on the original stream as a function ofat least a part of characteristics representative of the structure,content and parameters of the original stream, a target profile andexternal events.
 2. The process according to claim 1, wherein modifyingthe at least one block comprises replacing a part of the coefficients toproduce a main audio stream in nominal format and complementarymodification information that allows reconstruction of the originalstream by a decoder of target equipment, the scope of whichmodifications is variable and determined by the representativecharacteristics.
 3. The process according to claim 2, wherein themodified main stream is recorded on the target equipment prior totransmission of the complementary information to the target equipment.4. The process according to claim 2, wherein the modified main stream isrecorded on a physical support to be transmitted to the target equipmentprior to transmission of the complementary information to the targetequipment.
 5. The process according to claim 2, wherein the modifiedmain stream and the complementary information are transmitted togetherin real time.
 6. The process according to claim 2, wherein thecomplementary modification information comprises at least one digitalroutine suitable for executing a function.
 7. The process according toclaim 2, wherein the complementary modification information issubdivided into at least two subparts.
 8. The process according to claim7, wherein the subparts are distributed by different media.
 9. Theprocess according to claim 7, wherein the subparts are distributed bythe same media.
 10. The process according to claim 2, wherein thecomplementary information is transmitted on a physical vector.
 11. Theprocess according to claim 2, wherein the complementary information istransmitted online.
 12. The process according to claim 1, wherein thedigital audio sequences are modified in a differentiated manner as afunction of their audio content.
 13. The process according to claim 1,wherein digital audio sequences are modified in a differentiated manneras a function of the layer of modified scalability.
 14. The processaccording to claim 1, wherein digital audio sequences are modified in adifferentiated manner as a function of a rate in kilobits per second(kbits/s) of the original stream.
 15. The process according to claim 1,wherein the digital audio sequences are modified in a differentiatedmanner as a function of the target profile and of a digital leveldefined by a norm or standard with which they were encoded.
 16. Theprocess according to claim 1, wherein the digital audio sequences aremodified in a differentiated manner as a function of a number of audiochannels present in the stream.
 17. The process according to claim 1,wherein the digital audio sequences are modified in a differentiatedmanner as a function of coupling and multiplexing between differentaudio channels present in the stream.
 18. The process according to claim1, wherein the digital audio sequences are modified in a differentiatedmanner as a function of sampling frequency with which the audio streamwas encoded.
 19. The process according to claim 1, wherein the digitalaudio sequences are modified in a differentiated manner as a function ofa psychoacoustic model used.
 20. The process according to claim 1,wherein the digital audio sequences are modified in a differentiatedmanner as a function of granular scalability.
 21. The process accordingto claim 1, wherein the digital audio sequences are modified in adifferentiated manner as a function of an effect of desired auditorydegradation.
 22. The process according to claim 2, wherein the digitalaudio sequences are modified in a differentiated manner as a function ofa rate desired for the complementary information.
 23. The processaccording to claim 1, wherein values of subscripts of LSP's (“linespectral pairs”) or of subscripts of LSF's (“line spectral frequencies”)of the digital audio sequences are modified.
 24. The process accordingto claim 1, wherein values of delay and amplitude of a fundamental ofthe digital audio sequences are modified.
 25. The process according toclaim 1, wherein signs and position of excitation impulses of thedigital audio sequences are modified.
 26. The process according to claim1, wherein the modifications insure that authorized amplitude values ofsound are not exceeded.
 27. The process according to claim 2, whereinthe digital audio sequence reconstructed from the modified main streamand from the complementary information is auditorily identical to theoriginal stream, but different from a binary standpoint from theoriginal stream.
 28. The process according to claim 2, wherein thedigital audio sequence reconstructed from the modified main stream andfrom the complementary information is identical to the original streamauditorily and from a binary standpoint.
 29. The process according toclaim 1, wherein the digital audio sequences are modified in aprogressive manner increasing the degradation effect up to completescrambling of the audio stream.
 30. The process according to claim 1,wherein the digital audio sequences are modified with a randomgeneration of scrambling parameters and configurations.
 31. The processaccording to claim 1, further comprising an analog/digital conversionstage with a structured format applied to an analog audio signal.
 32. Asystem for distributing digital audio sequences comprising an audioserver comprising means for broadcasting a stream modified according toclaim 1 and a plurality of pieces of equipment provided with ascrambling circuit, wherein the server comprises means for recording thedigital profile of each target and means for control of the modificationmeans as a function of input variables corresponding to at least a partof characteristics representative of the structure, the content and theparameters of the original audio stream, the target profile and theexternal events.